[asterisk-dev] Asterisk 20.1.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Thu Dec 15 07:44:50 CST 2022
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 20.1.0.
This release candidate is available for immediate download at
https://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 20.1.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
Security bugs fixed in this release:
-----------------------------------
* ASTERISK-30338 - pjproject: Backport security fixes from
2.13
(Reported by Benjamin Keith Ford)
* ASTERISK-30176 - manager: GetConfig can read files outside of
Asterisk
(Reported by shawty)
* ASTERISK-30103 - chan_ooh323 Vulnerability in calling/called
party IE
(Reported by Michael Bradeen)
Improvements made in this release:
-----------------------------------
* ASTERISK-30328 - Typo in from_domain description on res_pjsip
configuration documentation
(Reported by Marcel Wagner)
* ASTERISK-30316 - res_pjsip: Documentation should point out
default if contact_user is not being set for outbound
registrations
(Reported by Marcel Wagner)
* ASTERISK-30289 - xmldoc: Allow XML docs to be reloaded
(Reported by N A)
* ASTERISK-30327 - rtp_engine.h: Remove obsolete example usage
(Reported by N A)
* ASTERISK-30286 - app_mixmonitor: Add option to use real
Caller ID for Caller ID
(Reported by N A)
* ASTERISK-30308 - pbx_builtins: Allow Answer to return
immediately
(Reported by N A)
* ASTERISK-30295 - test_json: Remove duplicated static
function
(Reported by N A)
* ASTERISK-30290 - file.c: Don't emit warnings on winks.
(Reported by N A)
* ASTERISK-30241 - res_pjsip_gelocation: Downgrade some NOTICE
scope trace debugs to DEBUG level
(Reported by N A)
* ASTERISK-30223 - features: add no-answer option to Bridge
application
(Reported by N A)
* ASTERISK-30158 - PJSIP: Add new 100rel option
"peer_supported"
(Reported by Maximilian Fridrich)
Bugs fixed in this release:
-----------------------------------
* ASTERISK-30349 - app_if: Format truncation error
(Reported by George Joseph)
* ASTERISK-30344 - ari: Memory leak in create when specifying
JSON
(Reported by Saken)
* ASTERISK-30283 - app_voicemail: Fix msg_create_from_file not
sending email to user
(Reported by N A)
* ASTERISK-30265 - res_pjsip_session: Fix missing PLAR support
on INVITEs
(Reported by N A)
* ASTERISK-29793 - adsi: CAS is malformed
(Reported by N
A)
* ASTERISK-30311 - func_presencestate: Fix invalid memory
access.
(Reported by N A)
* ASTERISK-30336 - sig_analog: Fix no timeout duration
(Reported by N A)
* ASTERISK-30244 - res_pjsip_pubsub: Occasional crash when
TCP/TLS connection terminated and subscription persistence is
removed
(Reported by nappsoft)
* ASTERISK-30184 - res_pjsip_session: re-INVITE after answering
results in wrong stream direction of first call leg
(Reported by Maximilian Fridrich)
* ASTERISK-29998 - sla: deadlock when calling SLAStation
application
(Reported by N A)
* ASTERISK-30321 - Build: Embedded blobs have executable
stacks
(Reported by George Joseph)
* ASTERISK-30293 - Memory leak in JSON_DECODE
(Reported
by David Uczen)
* ASTERISK-30314 - res_agi: RECORD FILE doesn't respect
"transmit_silence" asterisk.conf option
(Reported by
Joshua C. Colp)
* ASTERISK-30285 - manager.c: Remove outdated documentation
(Reported by N A)
* ASTERISK-30282 - CI: Coredump output isn't saved when running
unittests
(Reported by George Joseph)
* ASTERISK-30076 - app_stack: Incorrect exit location in
predial handlers logged
(Reported by N A)
* ASTERISK-30281 - chan_rtp: Local address being used before
being set
(Reported by George Joseph)
* ASTERISK-28689 - res_pjsip: Crash when locking group lock
when sending stateful response
(Reported by Jesse Ross)
* ASTERISK-30278 - tcptls: Abort occurs if SSL error is logged
if MALLOC_DEBUG is enabled
(Reported by N A)
* ASTERISK-30217 - Registration do not allow multiple proxies
(Reported by Igor Goncharovsky)
* ASTERISK-30273 - test_mwi: compilation fails on 32-bit
Debian
(Reported by N A)
* ASTERISK-30193 - chan_pjsip should return all codecs on a
re-INVITE without SDP
(Reported by Henning Westerholt)
* ASTERISK-30258 - Dialing API: Cancel a running async thread,
does not always cancel all calls
(Reported by Frederic LE
FOLL)
* ASTERISK-30274 - chan_dahdi: Unavailable channels are BUSY
(Reported by N A)
* ASTERISK-30264 - res_pjsip: Subscription handlers do not get
cleanly unregistered, causing crash
(Reported by N A)
* ASTERISK-30248 - ast_get_digit_str adds bogus initial
delimiter if first character not to be spoken
(Reported by
David Woolley)
* ASTERISK-30213 - Make crypto_load() reentrant and handle
symlinks correctly
(Reported by Philip Prindeville)
* ASTERISK-30256 - chan_dahdi: Fix format truncation warnings
(Reported by N A)
* ASTERISK-30239 - Prometheus plugin crashes Asterisk when
using local channel
(Reported by Joeran Vinzens)
* ASTERISK-30237 - res_prometheus: Crash when scraping bridges
(Reported by Igor Yeroshev)
* ASTERISK-30245 - db: ListItems is incorrect
(Reported
by N A)
* ASTERISK-30243 - func_logic: IF function complains if both
branches are empty
(Reported by N A)
* ASTERISK-30232 - Initialize stack-based ast_test_capture
structures correctly
(Reported by Philip Prindeville)
* ASTERISK-30220 - func_scramble: Fix segfault due to null
pointer deref
(Reported by N A)
* ASTERISK-30235 - res_crypto and tests: Memory issues and and
uninitialized variable error
(Reported by George Joseph)
* ASTERISK-30234 - res_geolocation: ...may be used
uninitialized error in geoloc_config.c
(Reported by George
Joseph)
* ASTERISK-30226 - REGRESSION: res_crypto complains about the
stir_shaken directory in /var/lib/asterisk/keys
(Reported
by George Joseph)
New Features made in this release:
-----------------------------------
* ASTERISK-21502 - New SIP Channel Driver - add Advice of
Charge support
(Reported by Matt Jordan)
* ASTERISK-30322 - res_hep: Add capture agent name support
(Reported by N A)
* ASTERISK-29497 - Add conditional branch applications
(Reported by N A)
* ASTERISK-30150 - res_pjsip_session: Add support for custom
parameters
(Reported by N A)
* ASTERISK-30305 - chan_dahdi: Allow FXO channels to start
immediately
(Reported by N A)
* ASTERISK-30284 - app_mixmonitor: Add option to delete
recording file when done
(Reported by N A)
* ASTERISK-30263 - res_pjsip_notify: Allow using
pjsip_notify.conf from AMI
(Reported by N A)
* ASTERISK-30146 - res_pjsip_logger: Add method-based log
filtering
(Reported by N A)
* ASTERISK-30091 - cdr: Allow CDRs to ignore call state
changes
(Reported by N A)
* ASTERISK-30254 - res_tonedetect: Add audible ringback
detection to TONE_DETECT
(Reported by N A)
* ASTERISK-30032 - Support of mediasec SIP headers and SDP
attributes
(Reported by Maximilian Fridrich)
* ASTERISK-30216 - app_bridgewait: Add option for BridgeWait to
not answer
(Reported by N A)
* ASTERISK-30179 - app_amd: Allow audio to be played while AMD
is running
(Reported by N A)
* ASTERISK-29432 - New function to allow access to any channel
(Reported by N A)
* ASTERISK-30222 - func_strings: Add trim functions
(Reported by N A)
For a full list of changes in this release candidate, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0-rc1
Thank you for your continued support of Asterisk!
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